all thanks for the replies. i know what to do now. thanks.John Novack [EMAIL PROTECTED] wrote: What provider?Pots lines?SOME providers will provide hunting on residential lines, but not all, and most probably not 4 lines. Hunting does not require any thing more than the providers switch
Yea, but even with that you can't run a macro from the CLI. You'd need the manager API to do this.
bp
On 8/30/06, Tim St. Pierre [EMAIL PROTECTED] wrote:
Sort of.There is a command line argument to the asterisk process that runsit's arguments as CLI commands.You could write a shell script that
Hello, I need some advice on the following problem I'm trying to solve:
At the office we are using 7940s as our phones, connected to an asterisk
box via SIP. Pretty standard setup, nothing fancy. Everyone has an
extension that comes out as a single line button on the phones, with the
second line
Hi,
In CLI, I can issue a dial command. How can I run a macro in CLI?
Is it possibe?
Thanks.
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I don't believe that the CLI was ever designed to do what you are asking. You should check out the manager API to do this type of stuff.
bp
On 8/30/06, unplug [EMAIL PROTECTED] wrote:
Hi,In CLI, I can issue a dial command.How can I run a macro in CLI?Is it possibe?Thanks.
I am new to asterisk. After studying the book and
the tutorial on the Internet, I am still confued about how to use context for
incoming calls.
Here is my question:
If I create s extensions in two different contexts
for incoming calls which one will be used? When a call comes in, which
gc wrote:
I am new to asterisk. After studying the book and the tutorial on the
Internet, I am still confued about how to use context for incoming calls.
Here is my question:
If I create s extensions in two different contexts for incoming calls
which one will be used? When a call comes in,
[EMAIL PROTECTED]
Date: Mon, 28 Aug 2006 07:40:34 -0700
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Question about context for incoming calls
I am new to asterisk. After studying the book and the tutorial on the
Internet, I am still confued about how to use context for incoming
Hi,
I have set a queue 5000 with a agent logged in. When an user
dial 5000, will ring and then answer the call. Queue function
will execute and recording is started. However, the recorded wav file
is not a valid file with just a few bytes. Anyone can help me to
enable the recording
When I put a call from an H323 phone to an asterisk box equiped with oh323
I cannot hear any ring tone on the phone (NetMeeting).
When call is answered everything is OK, i can hear and the other person too
can hear me.
I found this:
https://skylab.inaccessnetworks.com/mantis/view.php?id=79
Hi,
When I run iax2 show netstats... what is each side? Obviously Local
is me and Remote is the other side... but which is which direction?
That being is local me -- remote or is local remote -- me?
LOCAL -
REMOTE
Thanks! Actually, I want to share the asterisk DB using multiple
asterisks. So I use NFS to share the whole directory
/var/lib/asterisk in order to share files including astdb of asterisk.
However, there is not what I expected. Say, UA1 registers asterisk1
and UA2 registers asterisk2.
Unplug, I'm sure there are other people with better ideas but if you
see on sineapps, I remember someone having written a patch which
seperates out the the sip registry into a new table. If this is stable
and tested, then you might want to use that with an ARA configuration
and have all your
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
RR wrote:
Unplug, I'm sure there are other people with better ideas but if you
see on sineapps, I remember someone having written a patch which
seperates out the the sip registry into a new table. If this is stable
Save you searching:
Do you mean the patch can use to replace asterisk DB by ARA?
On 7/24/06, Matt Riddell (NZ) [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
RR wrote:
Unplug, I'm sure there are other people with better ideas but if you
see on sineapps, I remember someone having written
Hi,
What is the mechanism of asterisk DB? In folder /var/lib/asterisk,
there is a file called astdb. Does it used for storing data of
asterisk DB?
thanks,unplug
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PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of unplug
Sent: Friday, July 21, 2006 8:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] question about asterisk DB
Hi,
What is the mechanism of asterisk DB? In folder /var/lib/asterisk,
there is a file called
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of unplug
Sent: Friday, July 21, 2006 8:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] question about asterisk DB
Hi,
What is the mechanism of asterisk DB
That's right. We can find many information about Realtime (ARA). But
I can't find any information talking about asterisk DB. Where does it
store data? As I said before, there is a file
/var/lib/asterisk/astdb. Does it used to store data in asterisk DB?
Anymore information about asterisk DB
On Sat, 2006-07-22 at 11:34 +0800, unplug wrote:
That's right. We can find many information about Realtime (ARA). But
I can't find any information talking about asterisk DB. Where does it
store data? As I said before, there is a file
/var/lib/asterisk/astdb. Does it used to store data in
Kai Ober wrote:
Eric ManxPower Wieling schrieb:
Grandstream seems unable to produce stable firmware. They have tried
for *YEARS* and still people have to try many different versions of
the firmware to find one that actually works in their environment.
okay, i see, thx :)
i will try to
Hi,
I am using realtime function to query the table.
Table sipprop
name,number,forward
Peter,1234,0
May,4321,1
RealTime(sipprop|name|Peter) and it can simple get the output of
number if the record exist.
If I issue RealTime(sipprop|name|John), asterisk will show No
Realtime Matches Found.
Has somebody done that with a Grandstream GXP-2000 or a BudgetTone
100/101 ?
Has somebody even a list which SIP phones have this funtion?
Regards
Kai
It's called hotline or Private Line Auto Ringdown (PLAR).
SIP: It's a function of the phone, look for hotline in phone docs
Zap:
Kai Ober wrote:
Has somebody done that with a Grandstream GXP-2000 or a BudgetTone
100/101 ?
Has somebody even a list which SIP phones have this funtion?
SIPura supports it, Cisco ATAs support it. I assume that Cisco phones
support it.
I don't know about Grandstream devices since they
On 6/11/06, James Harper [EMAIL PROTECTED] wrote:
[.snip.]
My dialplan in the pap2 is:
(:0S0)
Which causes it to dial a '0' to asterisk as soon as I gets picked up.
In my asterisk dialplan it then does a DISA to another context, which
means Asterisk is doing all the dialplan stuff. For what I
Eric ManxPower Wieling schrieb:
I don't know about Grandstream devices since they are banned from our
network.
Banned? I didn't try any other devices, but whats wrong with the
Grandstreams??
wondering
Kai
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Kai Ober wrote:
Eric ManxPower Wieling schrieb:
I don't know about Grandstream devices since they are banned from our
network.
Banned? I didn't try any other devices, but whats wrong with the
Grandstreams??
Grandstream seems unable to produce stable firmware. They have tried
for *YEARS*
Eric ManxPower Wieling schrieb:
Grandstream seems unable to produce stable firmware. They have tried
for *YEARS* and still people have to try many different versions of
the firmware to find one that actually works in their environment.
okay, i see, thx :)
i will try to remember, if i'm
Hi,
I want to know about the content of ast db. It is like a registry
of the asterisk to store information about register users. The
similar user register information will be stored in DB in ARA. I want
to verify that when user sends a register request and it is valid,
asterisk will capture
Hi All:
I got inconsistent Agent complete events, I placed two calls to dial
the same queue 3666,
agent complete event has right information and time stamp in first
call. Second call was terminated at 09:24:59,
the agent complete event came in at wrong time at 09:24:42.
Does anyone know why?
Hi,
Does anyone know what asterisk do during the register invite
process using ARA (realtime) with Nat enabled?
Say there is an UA1 send an register request to asterisk. Asterisk
will parse the register request header (in which source file?). It
will get the necessary information update
I have a ring group set up with 3 extensions well
use 14, 15 and 16.
When a call comes in, it rings all three extensions. If one
particular extension already is on the phone, it completely skips that phone
and only rings the other 2. Example to explanation sake is:
Call comes in,
lf Of Chris
SuttonSent: Monday, June 26, 2006 11:15 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Question
about ring groups and ext. busy in call
I have a ring group set up with 3
extensions well use 14, 15 and 16.
When a call comes in, it rings all
three extensi
Does the CALLERID(all) also set the ANI Information, or
does the Set CALLERID(ani) also have to be called?
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All:
I tested echo test by dialing *43 under Asterisk configured by FreePbx
by using x-lite softphone. I could not figure out how the call is routed
to context from-internal. In sip_additional.conf, I have three
extensions defined as 2826, 2800 and 2801, which all are defined context
as
James Harper wrote:
Easy to do on the Linksys PAP2, if that helps. The functionality
probably depends on the make and model of the phone... maybe if you gave
those details as well?
James
Fantastic, this may solve the problem In the mail I've just posted
(which hasnt' appeared yet).
I
James Harper wrote:
Easy to do on the Linksys PAP2, if that helps. The functionality
probably depends on the make and model of the phone... maybe if you
gave
those details as well?
James
Fantastic, this may solve the problem In the mail I've just posted
(which hasnt' appeared yet).
On Sun, 2006-06-11 at 20:52 +1000, James Harper wrote:
Ideally I would have liked the pap2 to have done the same as 'immediate'
when talking about fxo, capi, misdn, etc, but I couldn't get it to
automatically dial nothing. A '0' was the best I could do. If anyone
knows how to put it into
On Sun, 2006-06-11 at 20:52 +1000, James Harper wrote:
Ideally I would have liked the pap2 to have done the same as
'immediate'
when talking about fxo, capi, misdn, etc, but I couldn't get it to
automatically dial nothing. A '0' was the best I could do. If anyone
knows how to put it into
James Harper wrote:
So... asterisk can't tell the difference between 's' for 'no extension
dialled', and when 's' was actually the name of the extension dialled...
is this the expected behaviour?
I surely hope so, you can refer to it as such in the extensions.conf as
well (with goto etc.)
Basically, I am looking to set up an extension which will be used as a
help-line.
I want it to function kind of like the bat phone from the old Batman series,
where Commissioner Gordon would pick up the extension in his office and it
would ring the phone back at Wayne's mansion. Is there a way
]
Sent: Sunday, 11 June 2006 10:42
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Question setting up a bat phone extension.
Basically, I am looking to set up an extension which will be used as a
help-line.
I want it to function kind of like the bat phone from the old Batman
It's called hotline or Private Line Auto Ringdown (PLAR).
SIP: It's a function of the phone, look for hotline in phone docs
Zap: immediate=yes, runs exten = when phone is picked up
Cisco and others: Look up PLAR
[EMAIL PROTECTED] wrote:
Basically, I am looking to set up an extension which will
Easy to do on the Linksys PAP2, if that helps. The functionality
probably
depends on the make and model of the phone... maybe if you gave
those details
as well?
James
Well, there's the rub. I don't have any of the
hardware yet. I am looking at the various options before buying anything.
Dear Friends and Supporters!
I have successfully installed the asterisk at home,
and it is working perfectly with sip and iax2 users. However, I am
wondering that should we use only sip or only iax2? If we use both
of them, does it take a lots of cpu resources to translate between sip
with
Hi there, SIP more commanly used and it is a openstandard. Meanwhile IAX2 is a protocol on asterisk. I dont think it will effect the cpu resources cause they are bid with the same codecs like G711 and etc..so if you used SIP or IAX2 it also refer to the same codecs...so dont think it will take a
Your Dial command must have both T and t in it to be able to transfer
both incoming and outgoing calls.
in features.conf, you can change # for blind transfer to ## -- this lets
you use # in banks and voicemails and other auto attendants.
Moj
Matt wrote:
Hi I'm a little confused here...
If I download zaptel-1.2.5, do I still have to apply the
zaptel-1.2.5-patch?
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Am Wednesday 26 April 2006 20:43 schrieb Wai Wu:
If I download zaptel-1.2.5, do I still have to apply the
zaptel-1.2.5-patch?
no.
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Hi,
If I am interfacing with a legacy PBX system a few questions.
#1 What do I need to do to configure 1 port on a dual port card as
pri_cpe and another as pri_net? Do I just change my config half-way
through the zaptel.conf file?
#2 When I setup span=1,1,0,esf,b8zs doesn't the esf
Hi All:
I used FreePBX to configure Asterisk, and tables are create in MySQL by
using FreePBX install script.
I created two x-lite softphone accounts by using FreePBX, they are
stored in table sip as friend.
I followed wiki doc to edit the extconfig.conf file.
I can not get those two softphone
Tielin Xu wrote:
I noticed that there is no ip address stored for my softphone in Mysql,
how does the Asterisk know which computer my softphone is running? I
checked the config files, no softphone registrations in sip.conf.
freePBX stores your phone information in sip_additional.conf and
Avi:
I got login failed message from x-lite.
I checked asterisk log, I got following:
res_config_mysql.c MySQL RealTime: Retrieve SQL: SELECT * from
sip_peers where name = '2826'
res_config_mysql.c MySQL RealTime: Everything is fine.
res_config_mysql.c MySQL RealTime: Failed to query database.
Tielin Xu wrote:
I'd like to use FreePBX, it seems some setup inconsistency with
Asterisk RealTime, do you know any other good admin tool for Asterisk?
FreePBX is not designed to work with Asterisk RealTime. I don't know of
a GUI to configure RealTime myself. :)
--
National Manager -
Hi I'm a little confused here... trying to setup a parking lot...
lot is setup but how do I send calls to the parkinglot? If I
allow #700 transfer, it seems I can only transfer on inbound calls...
if I use a T in my dialplan I can only transfer on outbound calls...
additionally pressing #
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Tuesday, April 11, 2006 8:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Question on clicking
This isn't an asterisk question, more of a telco question. Hopefully
someone on this list
This isn't an asterisk question, more of a telco question. Hopefully
someone on this list involved with telco can answer it.
I have a client who is using asterisk with analog phone lines. There
is a bad clicking noise on the line. Click Click Click.
We have narrowed it down to
Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Question on clicking
This isn't an asterisk question, more of a telco question. Hopefully
someone on this list involved with telco can answer it.
I have a client who is using asterisk with analog phone lines
their equipment...
Bob McDowell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Tuesday, April 11, 2006 8:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Question on clicking
This isn't
On Sonntag, 9. April 2006 9:58 Tele Cost Price Reducer wrote:
hi Ronald,
i would use a CallerIDNum authentication, based on the Asterisk
Database to solve it.
then you do not need any verification.
Dangerous. CallerIDs can be easily faked in some countries using VoIP
providers.
Kind
hi Ronald,
i would use a CallerIDNum authentication, based on the Asterisk Database to solve it.
then you do not need any verification.
you just build a list of approved numbers in the database and then have a context checking the whitelist.
if you need more help, let me know,
Mickey
On 4/8/06,
Lists,
Hi, good day, i was being task to create a DISA access for internal
purpose of the company, i'm having a problem to work with it with
authentication, but i think it's really a straight forward thing to do,
can someone enlight me on this. thanks
sample code snippet
exten =
Hi, I have questions about the Polycom 601 and side card
1) In the side card the lights all time off... But all functions it's ok.
I need help with extension module of polycom... All works fine... But lights
not work So... I don't know when any person or extension is busy...
Any ideas?
Augenstine
Sent: March 17, 2006 8:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Question about meetme app
My mistake. Locking a conference from the CLI does prevent any additional
callers from connecting. But AFAIK locking the conference does
I Would reccomend learning asterisk bit by bit. It's
the only way to do it. Here are several tools that you
can use.
1)Get [EMAIL PROTECTED] Its a great resource for
starters. You can learn a lot from it. You can get it
at http://asteriskathome.lists.digium.com
2)Read the book Asterisk, The future
2)Read the book Asterisk, The future of telephony.
You can buy it or download it for free. I dont have
the link to it but if some one else does please post
it.
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
___
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: Thursday, March 16, 2006 11:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Question about advanced IVR
Hi all,
I am a fairly new * user, so please bear with me. I have been scoring
the net for this info, but have had little success.
I am looking for any
I have a quick question about the MeetMe app. A locked conference means
what exactly?
A) That people can't join anymore
B) That everyone is muted except the admin
Follow-up question
If the answer above is A, how do you accomplish B?
Mick
___
A locked conference means that a pin number is required to join the
conference.
On Fri, 2006-03-17 at 16:20 -0500, Michael Gaudette wrote:
I have a quick question about the MeetMe app. A locked conference means
what exactly?
A) That people can't join anymore
B) That everyone is muted
, 2006 5:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Question about meetme app
A locked conference means that a pin number is required to join the
conference.
On Fri, 2006-03-17 at 16:20 -0500, Michael Gaudette wrote:
I have a quick question about
:[EMAIL PROTECTED] On Behalf Of Jonathan
Augenstine
Sent: March 17, 2006 5:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Question about meetme app
A locked conference means that a pin number is required to join the
conference.
On Fri, 2006-03
I have two simple questions regarding compiling Zaptel based on the
confusing (to a kernel newbie) instructions at
http://www.voip-info.org/wiki/view/Asterisk+Zaptel+Installation.
The instructions say:
For a Linux 2.6 kernel you may need to:
modify the EXTRAVERSION statement in Makefile so
Hi all,
I am a fairly new * user, so please bear with me. I have been scoring
the net for this info, but have had little success.
I am looking for any tips and info on how to build a information
delivery ivr app. By that I mean it will simply play prerecorded sound
files and respond to key
What exactly do you need? A digium card could be anything from one
pstn line, to multiple t1 lines, to who knows what else. And serial
number authentication...what's this for? Does a user dial in, enter
in a serial, then get access to something? Like a calling card, or
something completely
Hey all,
I have a client whose previous programmer ditched. I'm his webmaster,
and now he wants me to have an asterisk system set up for serial
number authentication...and I have a digium card from the previous
guy. Are there hosts that will set this up for me? ie, rack space
somwhere? Are
Hello,
This question is probabely recurrent, i apologize, but i haven't found a
limpid explanation (for me) in mail list, google, and hum source
code):
When use the Command DIAL to ring a group, WHERE is stored the name of
the 'winner' who pick up the call ? ($variable = ?), and, step
Hi,
We are using Asterisk 1.2.3 with RealTime for PSTN and Voicemail where
users register with an OpenSER cluster (2 nodes currently).
When they request PSTN they are forwarded to * where they have entries
in SIP realtime database. This ensures that they get their correct
CallerID and
Hello,
I have a situation where I need to differentiate between registrations
by users where there might be clashes on the left hand side (username)
portion of the SIP From URI. (for a multi-domain virtual hosting system)
It seems that only the username portion is used for SIP
My prior box would do this too, with exactly 99.987793 almost every time
as you saw. It only had a wildcard, x100p. actually, clone maybe. The
other notable difference is that I see you're using apic. have you
tried with the noapic kernel option to see if that changes your results?
On Mon, Jan 16, 2006 at 09:00:28PM -0500, Carlos Alperin wrote:
Another request make me test my t1 card, which has no quality problems, but
all that I get is:
[EMAIL PROTECTED] zaptel-1.2.1]# ./zttest
Opened pseudo zap interface, measuring accuracy...
99.987793% 99.987793%
Yes,
But I'm reading on the opposite side, or always I loose 1 sample?
[EMAIL PROTECTED] zaptel-1.2.1]# ./zttest -v
Opened pseudo zap interface, measuring accuracy...
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8191 sample intervals 99.987793%
8192 samples in 8191 sample
Another request make me test my t1 card, which has no
quality problems, but all that I get is:
[EMAIL PROTECTED] zaptel-1.2.1]# ./zttest
Opened pseudo zap interface, measuring accuracy...
99.987793% 99.987793% 99.987793% 99.987793% 99.987793%
99.987793% 99.987793%
99.987793%
Carlos Alperin wrote:
[EMAIL PROTECTED] zaptel-1.2.1]# ./zttest
Opened pseudo zap interface, measuring accuracy...
99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%
Is anything wrong about this?
I never get 100.00 % most of the times.
No problem there.
Hi,
I have about 53 SPA-2002 units out in the field. I've seen two or
three of them, now, exhibit an odd happening.
Users plug their phones into LINE1 (unless they have two lines). The
two users I've had issues with are both employees here who are fairly
knowledgeable in computers. They both
Could be they pressed the DND code for the SPA (don't remember by
heart what it is, something like *xx). The easiest way to check is to
log into the http server of the SPA and check the status on the first
page.
On 1/3/06, Matt [EMAIL PROTECTED] wrote:
Hi,
I have about 53 SPA-2002 units out in
That's possible, and I didn't think about that.(I wil check). however
I did totally wipe the configuration on the device *** RESET and then
reprogrammed it and the same problem happened, so I kind of doubt that
was the issue.
On 1/3/06, C F [EMAIL PROTECTED] wrote:
Could be they pressed the DND
.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt
Sent: Tuesday, January 03, 2006 8:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Question on SPA-2002
That's possible, and I didn't think about
If I'm in a meetme conference, what would I need to do to have some
call files make calls and connect them into the meetme conference with
me?
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I'm not sure if Priority is the correct term, but it is the order number
as in exten = fax,1-
If I have an application that loads / includes another file, will a line
of the same order in the included file override the one in the main
application? What I need to do is:
[test]
include
Hello All,
I am using * 1.2, BRIstuff 0.3 PRE1, Dual HFC-PCI, 1x TE, 1x NT
I am using DECT phones on a Siemens ISDN phone/DECT-base.
My dial options are rTtWw, automon=*1, blindxfer=##
Whether I am calling (to my cell) or being called (from my cell), only the
caller can initiate recording or
Kevin Bockman wrote:
This is just a feature of PRI service. Of course all of the call info
is still available even if you 'block' it. The call still has to be
traceable. Magic huh? I thought that was cool too the first time I
found out about it.
It depends on whether you are purchasing
Matt wrote:
For instance, when someone blocks their number it comes into our
system with the block flag (across PRI). It is then passed on to the
ATA as blocked. Is it legal for me to set the flag back to unblock
the call? (I realize no one here is probably a lawer but was just
curious to
On Oct 26, 2005, at 10:31 PM, Kevin P. Fleming wrote:
Matt wrote:
For instance, when someone blocks their number it comes into our
system with the block flag (across PRI). It is then passed on to
the
ATA as blocked. Is it legal for me to set the flag back to
unblock
the call? (I
Dave Grey wrote:
I found this pretty interesting -- I didn't know that when you blocked
CID with *67 or per-line blocking that it went anywhere at all.
Apparently, IPKall (who I am using for DID) is doing this
unblocking. I tested a couple of different numbers (A cellphone on a
network
Hi,
Does anyone know what the legalness is of unblocking a blocked call?
For instance, when someone blocks their number it comes into our
system with the block flag (across PRI). It is then passed on to the
ATA as blocked. Is it legal for me to set the flag back to unblock
the call? (I
Hello all. I
am new to Asterisk as well as this group so please excuse me for a bit as I
learn the
ropes of
Asterisk. Anyway, I currently am using a pap2-na adapter with Teliax
and Mesa Networks (my isp) and
was wondering what I
will need to get Asterisk running correctly. I am wondering
PROTECTED] On Behalf Of Steve Totaro
Sent: Tuesday, October 04, 2005
12:21 AM
To: Jorge Cisneros; Asterisk Users
Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Question about 3Com(r) 3101 Basic Phone
I have no idea about the 3XXX series of phones. The
2XXX used
Valentine
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion' ; 'Jorge Cisneros'
Sent: Wednesday, October 05, 2005 12:20
PM
Subject: RE: [Asterisk-Users] Question
about 3Com(r) 3101 Basic Phone
The 3101 is in the
same boat as the rest of the 31xx and 2102B/PE
Hi, i have one question, the 3Com 3101 Basic Phone work with
asterisk, if so i any a especial firmware o another thing. And wath
other 3com ip phone product work with asterisk. I think is a good idea
to create a list with the all voip device and the status with asterisk.
I have no idea about the 3XXX series of
phones. The 2XXX used to have SIP firmware but I could never get my hands
on it. I used to see the SIP 2XXX phones selling on Ebay from time to
time. I imagine that even if you can locate the SIP firmware for the old
phones, you would have to upload
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